rendered paste body<--- SIP read from 192.168.1.142:5060 --->
INVITE sip:914805775268@192.168.1.20;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.142;branch=z9hG4bK3502d7e8274D2553
From: "Dan Trainor" <sip:101@192.168.1.20>;tag=30F10D6-94944C67
To: <sip:914805775268@192.168.1.20;user=phone>
CSeq: 1 INVITE
Call-ID: 412b3a12-499a898c-88b504d5@192.168.1.142
Contact: <sip:101@192.168.1.142>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.2.0041
Supported: 100rel,replace
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 249
v=0
o=- 978308813 978308813 IN IP4 192.168.1.142
s=Polycom IP Phone
c=IN IP4 192.168.1.142
t=0 0
a=sendrecv
m=audio 2260 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
<------------->
--- (14 headers 11 lines) ---
Sending to 192.168.1.142 : 5060 (no NAT)
Using INVITE request as basis request - 412b3a12-499a898c-88b504d5@192.168.1.142
Found peer '101'
<--- Reliably Transmitting (no NAT) to 192.168.1.142:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.142;branch=z9hG4bK3502d7e8274D2553;received=192.168.1.142
From: "Dan Trainor" <sip:101@192.168.1.20>;tag=30F10D6-94944C67
To: <sip:914805775268@192.168.1.20;user=phone>;tag=as1018a264
Call-ID: 412b3a12-499a898c-88b504d5@192.168.1.142
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="26463511"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '412b3a12-499a898c-88b504d5@192.168.1.142' in 32000 ms (Method: INVITE)
pbx*CLI>
<--- SIP read from 192.168.1.142:5060 --->
ACK sip:914805775268@192.168.1.20 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.142;branch=z9hG4bK3502d7e8274D2553
From: "Dan Trainor" <sip:101@192.168.1.20>;tag=30F10D6-94944C67
To: <sip:914805775268@192.168.1.20;user=phone>;tag=as1018a264
CSeq: 1 ACK
Call-ID: 412b3a12-499a898c-88b504d5@192.168.1.142
Contact: <sip:101@192.168.1.142>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.2.0041
Max-Forwards: 70
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
pbx*CLI>
<--- SIP read from 192.168.1.142:5060 --->
INVITE sip:914805775268@192.168.1.20;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.142;branch=z9hG4bK1319c891FB06E70
From: "Dan Trainor" <sip:101@192.168.1.20>;tag=30F10D6-94944C67
To: <sip:914805775268@192.168.1.20;user=phone>
CSeq: 2 INVITE
Call-ID: 412b3a12-499a898c-88b504d5@192.168.1.142
Contact: <sip:101@192.168.1.142>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.2.0041
Supported: 100rel,replace
Allow-Events: talk,hold,conference
Proxy-Authorization: Digest username="101", realm="asterisk", nonce="26463511", uri="sip:914805775268@192.168.1.20;user=phone", response="08520023e3c026fac24f529a5e46879a", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 249
v=0
o=- 978308813 978308813 IN IP4 192.168.1.142
s=Polycom IP Phone
c=IN IP4 192.168.1.142
t=0 0
a=sendrecv
m=audio 2260 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
<------------->
--- (15 headers 11 lines) ---
Sending to 192.168.1.142 : 5060 (no NAT)
Using INVITE request as basis request - 412b3a12-499a898c-88b504d5@192.168.1.142
Found peer '101'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.142:2260
Found description format PCMU for ID 0
Found description format PCMA for ID 8
Found description format G729 for ID 18
Found description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.142:2260
Looking for 914805775268 in numberplan-custom-1 (domain 192.168.1.20)
<--- Reliably Transmitting (no NAT) to 192.168.1.142:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.142;branch=z9hG4bK1319c891FB06E70;received=192.168.1.142
From: "Dan Trainor" <sip:101@192.168.1.20>;tag=30F10D6-94944C67
To: <sip:914805775268@192.168.1.20;user=phone>;tag=as1018a264
Call-ID: 412b3a12-499a898c-88b504d5@192.168.1.142
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '412b3a12-499a898c-88b504d5@192.168.1.142' in 32000 ms (Method: INVITE)
pbx*CLI>
<--- SIP read from 192.168.1.142:5060 --->
ACK sip:914805775268@192.168.1.20 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.142;branch=z9hG4bK1319c891FB06E70
From: "Dan Trainor" <sip:101@192.168.1.20>;tag=30F10D6-94944C67
To: <sip:914805775268@192.168.1.20;user=phone>;tag=as1018a264
CSeq: 2 ACK
Call-ID: 412b3a12-499a898c-88b504d5@192.168.1.142
Contact: <sip:101@192.168.1.142>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.2.0041
Max-Forwards: 70
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
pbx*CLI>